Audio conference platform with dynamic speech detection threshold

ABSTRACT

The present invention comprises a method for audio/video conferencing. In a preferred embodiment, the method comprises using a dynamic threshold value to determine whether there is speech on a line. One aspect, the method comprises determining a dynamic threshold value based on one or more characteristics of signals received on a port, associating that dynamic threshold value with the port; and comparing one or more characteristics of signals subsequently received on the port to the dynamic threshold value. Signals received over a plurality of ports are summed, but for ports whose signal characteristics have a specified relationship to the dynamic threshold value associated with that port, signals are not contained in the sum.

CROSS-REFERENCE TO RELATED APPLICATIONS

[0001] This application claims priority to U.S. Provisional ApplicationNo. 60/287,441, filed Apr. 30, 2001, the entire contents of which areincorporated herein by reference.

BACKGROUND OF THE INVENTION

[0002] The present invention relates to telephony, and in particular toan audio conferencing platform.

[0003] Audio conferencing platforms are known. For example, see U.S.Pat. Nos. 5,483,588 and 5,495,522. Audio conferencing platforms allowconference participants to easily schedule and conduct audio conferenceswith a large number of users. In addition, audio conference platformsare generally capable of simultaneously supporting many conferences.

[0004] A problem with existing audio conference platforms is that theyemploy a fixed threshold to determine whether a conference participantis speaking. Using such a fixed threshold may result in a conferenceparticipant being added to the summed conference audio, even though theyare not speaking. Specifically, if the background audio noise is high(e.g., the user is on a factory floor), then the amount of digitizedaudio energy associated with that conference participant may besufficient for the conference platform to falsely detect speech, and addthe background noise to the conference sum under the mistaken beliefthat the energy is associated with speech.

[0005] Therefore, there is a need for a system that accounts forbackground noise in the detection of valid conference speakers.

SUMMARY OF THE INVENTION

[0006] One object of the present invention is to provide a method andsystem that advantageously accounts for background noise on linesparticipating in a conference call and prevents the background noisefrom being added to the conference sum because an erroneousdetermination has been made that the energy is associated with speech.Another object is to provide such an advantage dynamically, to accountfor changing conditions on participating lines.

[0007] A preferred embodiment of the invention comprises an audioconferencing platform that includes a time division multiplexing (TDM)data bus, a controller, and an interface circuit that receives audiosignals from a plurality of conference participants and providesdigitized audio signals in assigned time slots over the data bus. Theaudio conferencing platform also includes a plurality of digital signalprocessors (DSPs) adapted to communicate on the TDM bus with theinterface circuit. At least one of the DSPs sums a plurality of thedigitized audio signals associated with conference participants who arespeaking to provide a summed conference signal. This DSP provides thesummed conference signal to at least one of the other plurality of DSPs,which removes the digitized audio signal associated with a speaker whosevoice is included in the summed conference signal, thus providing acustomized conference audio signal to each of the speakers.

[0008] Each of the digitized audio signals are processed to determinewhether the digitized audio signal includes speech. For each digitizedaudio signal, the amount of energy associated with the digitized audiosignal is compared against a dynamic threshold value associated with theline over which the audio signal is received. The dynamic thresholdvalue is set as a function of background noise within the digitizedaudio signal.

[0009] The audio conferencing platform preferably configures at leastone of the DSPs as a centralized audio mixer and at least another one ofthe DSPs as an audio processor. The centralized audio mixer performs thestep of summing a plurality of the digitized audio signals associatedwith conference participants who are speaking, to provide the summedconference signal. The centralized audio mixer provides the summedconference signal to the audio processor(s) for post processing androuting to the conference participants. The post processing includesremoving the audio associated with a speaker from the conference signalto be sent to the speaker. For example, if there are forty conferenceparticipants and three of the participants are speaking, then the summedconference signal will include the audio from the three speakers. Thesummed conference signal is made available on the data bus to thethirty-seven non-speaking conference participants. However, the threespeakers each receive an audio signal that is equal to the summedconference signal less the digitized audio signal associated with thatspeaker. Removing the speaker's own voice from the audio he hearsreduces echoes.

[0010] The centralized audio mixer also preferably receives DTMF detectbits indicative of the digitized audio signals that include a DTMF tone.The DTMF detect bits may be provided by another of the DSPs that isprogrammed to detect DTMF tones. If the digitized audio signal isassociated with a speaker, but the digitized audio signal includes aDTMF tone, the centralized conference mixer will not include thedigitized audio signal in the summed conference signal while that DTMFdetect bit signal is active. This ensures that conference participantsdo not hear annoying DTMF tones in the conference audio. When the DTMFtone is no longer present in the digitized audio signal, the centralizedconference mixer may include the audio signal in the summed conferencesignal.

[0011] The audio conference platform is preferably capable of supportinga number of simultaneous conferences (e.g., 384). As a result, the audioconference mixer provides a summed conference signal for each of theconferences.

[0012] Each of the digitized audio signals may be preprocessed. Thepreprocessing steps include decompressing the signal (e.g., using thewell-known μ-law or A-law compression schemes), and determining whetherthe magnitude of the decompressed audio signal is greater than adetection threshold. If it is, then a speech bit associated with thedigitized audio signal is set. Otherwise, the speech bit is cleared.

[0013] The centralized conference mixer reduces repetitive tasksdistributed between the plurality of DSPs. In addition, centralizedconference mixing provides a system architecture that is scalable andthus easily expanded.

[0014] Advantageously, using a dynamic threshold value to determinewhether there is speech on a line helps to ensure that background noiseis not falsely detected as speech.

[0015] Thus, a method in accordance with a preferred embodiment of thepresent invention comprises receiving audio signals over a plurality ofports. For at least one port, the method comprises determining a dynamicthreshold value based on one or more characteristics of signals receivedon the port; associating said dynamic threshold value with the port; andcomparing one or more characteristics of signals subsequently receivedon the port to the dynamic threshold value. The method further comprisessumming signals received over the plurality of ports, wherein signalsreceived on the at least one port whose characteristics (such as energylevel) have a specified relationship to the dynamic threshold value (forexample, having an energy level less than the threshold value) are notcontained in the sum. The method may further comprise preprocessingaudio signals by decompressing them using either μ-law or A-lawdecompression.

[0016] In one aspect, the method comprises identifying which ports arereceiving audio signals that contain speech; and, on each suchidentified port, transmitting a summed signal, wherein said summedsignal does not contain signals received on that port.

[0017] In another aspect, the method comprises identifying which portsare receiving audio signals that contain DTMF tones; and, on each suchidentified port, transmitting a summed signal, wherein said summedsignal does not contain signals received on that port. Preferably, thestep of identifying comprises setting a DTMF detect bit for a signal.The method may also comprise the step of including signals frompreviously identified ports in the sum after those ports are no longeridentified as receiving signals containing one or more DTMF tones.

[0018] The invention further comprises software and systems forimplementing methods described herein.

[0019] These and other objects, features, and advantages of the presentinvention will become apparent in light of the following detaileddescription of preferred embodiments thereof, as illustrated in theaccompanying drawings.

[0020] Although the invention has been described in connection with anaudio conferencing platform, it is not limited to such a platform andmay be used, for example, in a video conferencing system.

BRIEF DESCRIPTION OF THE DRAWINGS

[0021]FIG. 1 illustrates a conferencing system in accordance with apreferred embodiment of the present invention;

[0022]FIG. 2 illustrates a functional block diagram of an audioconferencing platform of a preferred embodiment within the conferencingsystem of FIG. 1;

[0023]FIG. 3 is a block diagram illustration of a processor board of apreferred embodiment within the audio conferencing platform of FIG. 2;

[0024]FIG. 4 is a functional block diagram illustration of resources onthe processor board of FIG. 3;

[0025]FIG. 5 is a flow chart illustrating the processing of signalsreceived from network interface cards over a TDM bus;

[0026]FIG. 6 is a flow chart illustration of the DTMF tone detectionprocessing;

[0027]FIGS. 7A-7B together provide a flow chart illustration ofpreferred conference mixer processing to create a summed conferencesignal; and

[0028]FIG. 8 is a flow chart illustrating the processing of signals tobe output to the network interface cards via the TDM bus.

DETAILED DESCRIPTION OF THE INVENTION

[0029]FIG. 1 is a diagram of a conferencing system 20 in accordance witha preferred embodiment of the present invention. The system 20 connectsa plurality of user sites 21-23 through a switching network 24 to anaudio conferencing platform 26. The plurality of user sites may bedistributed worldwide, or at a company facility/campus. For example,each of the user sites 21-23 may be in different cities and connected tothe audio platform 26 via the switching network 24, which may includePSTN and PBX systems. The connections between the user sites and theswitching network 24 may include T1, E1, T3, and ISDN lines.

[0030] Each user site 21-23 preferably includes one or more telephones28 and one or more personal computers or servers 30. However, a usersite may only include either a telephone, such as user site 21 a, or acomputer/server, such as user site 23 a. The computer/server 30 may beconnected via an Intemet/intranet backbone 32 to a server 34. The audioconferencing platform 26 and the server 34 are connected via a data link36 (e.g., a {fraction (10/100)} Base T Ethernet link). The computer 30allows the user to participate in a data conference simultaneous to theaudio conference via the server 34. In addition, the user can use thecomputer 30 to interface (e.g., via a browser) with the server 34 toperform functions such as conference control, administration (e.g.,system configuration, billing, reports, . . . ), scheduling and accountmaintenance. The telephone 28 and the computer 30 may cooperate toprovide voice over the Internet/intranet 32 to the audio conferencingplatform 26 via the data link 36.

[0031]FIG. 2 is a functional block diagram of an audio conferencingplatform 26 in accordance with a preferred embodiment of the presentinvention. The audio conferencing platform 26 includes a plurality ofnetwork interface cards (NICs) 38-40 that receive audio information fromthe switching network 24 (see FIG. 1). Each NIC is preferably capable ofhandling a plurality of different trunk lines (e.g., eight). The datareceived by the NIC is generally an 8-bit μ-law or A-law sample. The NICplaces the sample into a memory device (not shown), which is used tooutput the audio data onto a data bus. The data bus is preferably a TDMbus based, in one embodiment, upon the H.110 telephony standard.

[0032] The audio conferencing platform 26 also includes a plurality ofprocessor boards 44-46 that receive and transmit data to the NICs 38-40over the TDM bus 42. The NICs and the processor boards 44-46 alsocommunicate with a controller/CPU board 48 over a system bus 50. Thesystem bus 50 is preferably based upon the Compact Peripheral ComponentInterconnect (“cPCI”) standard. The CPU/controller communicates with theserver 34 (see FIG. 1) via the data link 36. The controller/CPU boardmay include a general purpose processor such as a 200 MHz Pentium™CPUmanufactured by Intel Corporation, a processor from AMD or any othersimilar processor (including an ASIC) having sufficient processor speed(MIPS) to support the present invention.

[0033]FIG. 3 is block diagram illustration of the processor board 44.The board 44 includes a plurality of dynamically programmable digitalsignal processors 60-65. Each digital signal processor (DSP) is anintegrated circuit that communicates with the controller/CPU card 48(see FIG. 2) over the system bus 50. Specifically, the processor board44 includes a bus interface 68 that interconnects the DSPs 60-65 to thesystem bus 50. Each DSP also includes an associated dual port RAM (DPR)70-75 that buffers commands and data for transmission between the systembus 50 and the associated DSP.

[0034] Each DSP 60-65 also transmits data over and receives data fromthe TDM bus 42. The processor card 44 includes a TDM bus interface 78that performs any necessary signal conditioning and transformation. Forexample, if the TDM bus is an H.110 bus, it includes thirty-two seriallines. As a result the TDM bus interface may include aserial-to-parallel and a parallel-to-serial interface.

[0035] Each DSP 60-65 also includes an associated TDM dual port RAM80-85 that buffers data for transmission between the TDM bus 42 and theassociated DSP.

[0036] Each of the DSPs is preferably a general purpose digital signalprocessor IC, such as the model number TMS320C6201 processor availablefrom Texas Instruments. The number of DSPs resident on the processorboard 44 is a function of the size of the integrated circuits, theirpower consumption, and the heat dissipation ability of the processorboard. For example, in certain embodiments there may be between four andten DSPs per processor board.

[0037] Executable software applications may be downloaded from thecontroller/CPU 48 (see FIG. 2) via the system bus 50 to a selectedone(s) of the DSPs 60-65. Each of the DSPs is preferably also connectedto an adjacent DSP via a serial data link.

[0038]FIG. 4 is illustrates the DSP resources on the processor board 44illustrated in FIG. 3. Referring to FIGS. 3 and 4, the controller/CPU 48(see FIG. 2) downloads executable program instructions to a DSP basedupon the function that the controller/CPU assigns to the DSP. Forexample, the controller/CPU may download executable program instructionsfor the DSP3 62 to function as an audio conference mixer 90, while theDSP2 61 and the DSP4 63 may be configured as audio processors 92, 94,respectively. Other DSPs 60, 65 may be configured by the controller/CPU48 (see FIG. 2) to provide services such as DTMF detection 96, audiomessage generation 98 and music playback 100.

[0039] Each audio processor 92, 94 is capable of supporting a certainnumber of user ports (i.e., conference participants). This number isbased upon the operational speed of the various components within theprocessor board and the over-all design of the system. Each audioprocessor 92, 94 receives compressed audio data 102 from the conferenceparticipants over the TDM bus 42.

[0040] The TDM bus 42 may, for example, support 4096 time slots, eachhaving a bandwidth of 64 kbps. The timeslots are generally dynamicallyassigned by the controller/CPU 48 (see FIG. 2) as needed for theconferences that are currently occurring. However, one of ordinary skillin the art will recognize that in a static system the timeslots may bepredetermined.

[0041]FIG. 5 is a flow chart illustrating the processing steps 500performed by each audio processor on the digitized audio signalsreceived over the TDM bus 42 from the NICs 38-40 (see FIG. 2). Theexecutable program instructions associated with these processing steps500 are typically downloaded to the audio processors 92, 94 (see FIG. 4)by the controller/CPU 48 (see FIG. 2). The download may occur duringsystem initialization or reconfiguration. These processing steps 500preferably are executed at least once every 125 microseconds to provideaudio of the requisite quality.

[0042] For each of the active/assigned ports for the audio processor,step 502 reads the audio data for that port from TDM dual port RAMassociated with the audio processor. For example, if DSP2 61 (see FIG.3) is configured to perform the function of audio processor 92 (see FIG.4), then the data is read from the read bank of the TDM dual port RAM81. If the audio processor 92 is responsible for, for example, 700active/assigned ports, then step 502 reads the 700 bytes of associatedaudio data from the TDM dual port RAM 81. Each audio processor includesa time slot allocation table (not shown) that specifies the addresslocation in the TDM dual port RAM for the audio data from each port.

[0043] Since each of the audio signals is typically compressed (e.g.,μ-law, A-law), step 504 decompresses each of the 8-bit signals to a16-bit word. Step 506 computes the average magnitude (AVM) for each ofthe decompressed signals associated with the ports assigned to the audioprocessor. For additional details, see co-pending U.S. patentapplication Ser. No. 09/532,602, filed Mar. 22, 2000, entitled “ScalableAudio Conference Platform,” the entire contents of which areincorporated herein by reference for all purposes.

[0044] Step 508 is performed to determine which of the ports arespeaking. This step compares the average magnitude for the port computedin step 506 against a predetermined magnitude value representative ofspeech (e.g., −35 dBm). If average magnitude for the port exceeds thepredetermined magnitude value representative of speech, a speech bitassociated with the port is set. Otherwise, the associated speech bit iscleared. Each port has an associated speech bit. Step 510 outputs allthe speech bits (eight per timeslot) onto the TDM bus. Step 512 isperformed to calculate an automatic gain correction (AGC) value for eachport. To compute an AGC value for the port, the AVM value is convertedto an index value associated with a table containing gain/attenuationfactors. For example, there may be 256 index values, each uniquelyassociated with 256 gain/attenuation factors. The index value is used bythe conference mixer 90 (see FIG. 4) to determine the gain/attenuationfactor to be applied to an audio signal that will be summed to createthe conference sum signal.

[0045] In a preferred embodiment, the threshold used in step 508 todetermine whether speech is present is a dynamic speech detectionthreshold value, set as a function of the noise detected on the line.For example, if the magnitude for the energy for the line/port exceeds anoise detection threshold value for a predetermined amount of time(e.g., three seconds), then noise is detected and a higher thresholdvalue may be used in step 510 to determine whether the user is speaking.Once noise has been detected, the dynamic threshold value may be set asa function of the magnitude of the energy on the line. For example, thedynamic threshold value may be set to a certain value greater than thevalue of the noise on the line (e.g., the average noise). Each line mayemploy a different speech detection threshold, since the backgroundnoise on each of the lines may be different.

[0046] The system may also set a noise bit for the line, and the noisebit may be provided to the controller/CPU 48 (see FIG. 2) to take thenecessary action due to the background noise. The action may include notallowing this conference participant to be on the speech list (i.e., thelist of lines summed to create the conference signal), or sending anaudio message to the conference participant that the system detects highbackground noise and recommends that the conference participant try totake corrective action (e.g., move to a different area, close an officedoor, go off speaker phone, etc.).

[0047] Additional action may include sending an audio message to theconference participant that the system detects high background noise andinstructing the participant to hit a key on the telephone keypad so thesystem does not consider the audio from the participant for theconference audio. The system would then detect the DTMF tone associatedwith the key being depressed and take the necessary action to preventaudio from this participant from being used in the conference sum, untilsuch time that the user, for example, hits the same key again or anotherkey instructing the system to consider audio from the participant forthe conference sum.

[0048]FIG. 6 is a flow chart illustration of the DTMF tone detectionprocessing 600. These processing steps 600 are performed by the DTMFprocessor 96 (see FIG. 4), preferably at least once every 125microseconds, to detect DTMF tones within digitized audio signals fromthe NICs 38-40 (FIG. 2). One or more of the DSPs may be configured tooperate as a DTMF tone detector. The executable program instructionsassociated with the processing steps 600 are typically downloaded by thecontroller/CPU 48 (see FIG. 2) to the DSP designated to perform the DTMFtone detection function. The download may occur during initialization orsystem reconfiguration.

[0049] For an assigned number of the active/assigned ports of theconferencing system, step 602 reads the audio data for the port from theTDM dual port RAM associated with the DSP(s) configured to perform theDTMF tone detection function. Step 604 then expands the 8-bit signal toa 16-bit word. Next, step 606 tests each of these decompressed audiosignals to determine whether any of the signals includes a DTMF tone.For any signal that does include a DTMF tone, step 606 sets a DTMFdetect bit associated with the port. Otherwise, the DTMF detect bit iscleared. Each port has an associated DTMF detect bit. Step 608 informsthe controller/CPU 48 (see FIG. 3) through Dual Port Ram (DPR) whichDTMF tone was detected, since the tone is representative of systemcommands and/or data from a conference participant. Step 610 outputs theDTMF detect bits onto the TDM bus.

[0050]FIGS. 7A-7B collectively provide a flow chart illustratingprocessing steps 700 performed by the audio conference mixer 90 (seeFIG. 4), preferably at least once every 125 microseconds, to create asummed conference signal for each conference. The executable programinstructions associated with the processing steps 700 are typicallydownloaded by the controller/CPU 48 (see FIG. 2) over the system bus 50(see FIG. 2) to the DSP designated to perform the conference mixerfunction. The download may occur during initialization or systemreconfiguration.

[0051] Referring to FIG. 7A, for each of the active/assigned ports ofthe audio conferencing system, step 702 reads the speech bit and theDTMF detect bit received over the TDM bus 42 (see FIG. 4).Alternatively, the speech bits may be provided over a dedicated seriallink that interconnects the audio processor or processorsand theconference mixer. Step 704 is then performed to determine whether thespeech bit for the port is set (i.e., whether energy that may be speechis detected on that port). If the speech bit is set, then step 706 isperformed to see whether the DTMF detect bit for the port is also set.If the DTMF detect bit is clear, then the audio received by the port isspeech and the audio does not include DTMF tones. As a result, step 708sets the conference bit for that port; otherwise, step 709 clears theconference bit associated with the port. Since the audio conferencingplatform 26 (see FIG. 1) preferably can support many simultaneousconferences (e.g., 384), the controller/CPU 48 (see FIG. 2) keeps trackof the conference that each port is assigned to and provides thatinformation to the DSP performing the audio conference mixer function.Upon the completion of step 708, the conference bit for each port hasbeen updated to indicate the conference participants whose voice shouldbe included in the conference sum.

[0052] Referring to FIG. 7B, for each of the conferences, step 710 isperformed, if needed, to decompress each of the audio signals associatedwith conference bits that are set. Step 711 performs AGC and gain/TLP(Test Level Point) compensation on the expanded signals from step 710.Step 712 is then performed to sum each of the compensated audio samplesto provide a summed conference signal. Since many conferenceparticipants may be speaking at the same time, the system preferablylimits the number of conference participants whose voice is summed tocreate the conference audio. For example, the system may sum the audiosignals from a maximum of three speaking conference participants. Step714 outputs the summed audio signal for the conference to the audioprocessors, as appropriate. In a preferred embodiment, the summed audiosignal for each conference is output to the audio processor(s) over theTDM bus. Since the audio conferencing platform supports a number ofsimultaneous conferences, steps 710-714 are performed for each of theconferences.

[0053]FIG. 8 is a flow chart illustrating the processing steps 800performed by each audio processor to output audio signals over the TDMbus to conference participants. The executable program instructionsassociated with these processing steps 800 are typically downloaded toeach audio processor by the controller/CPU during system initializationor reconfiguration. These steps 800 are also preferably executed atleast once every 125 microseconds.

[0054] For each active/assigned port, step 802 retrieves the summedconference signal for the conference that the port is assigned to. Step804 reads the conference bit associated with the port, and step 806tests the bit to determine whether audio from the port was used tocreate the summed conference signal. If it was, then step 808 removesthe gain (e.g., AGC and gain/TLP) compensated audio signal associatedwith the port from the summed audio signal. This step removes thespeaker's own voice from the conference audio. If step 806 determinesthat audio from the port was not used to create the summed conferencesignal, then step 808 is bypassed. To prepare the signal to be output,step 810 applies a gain, and step 812 compresses the gain correctedsignal. Step 814 then outputs the compressed signal onto the TDM bus forrouting to the conference participant associated with the port, via theNIC (see FIG. 2).

[0055] Preferably, the audio conferencing platform 26 (see FIG. 1)computes conference sums at a central location. This reduces thedistributed summing that would otherwise need to be performed to ensurethat the ports receive the proper conference audio. In addition, theconference platform is readily expandable by adding additional NICsand/or processor boards. That is, the centralized conference mixerarchitecture allows the audio conferencing platform to be scaled to theuser's requirements.

[0056] One of ordinary skill will appreciate that the overall systemdesign is a function of the processing ability of each DSP. For example,if a sufficiently fast DSP is available, then the functions of the audioconference mixer, the audio processor and the DTMF tone detection andthe other DSP functions may be performed by a single DSP.

[0057] In addition, although the aspect of the dynamic threshold valuehas been discussed in the context of a system that employs a centralizedsumming architecture, one of ordinary skill in the art will recognizethat dynamic thresholding is certainly not limited to systems with acentralized summing architecture. It is contemplated that all audioconferencing systems, and systems with similar audio cpabailities, wouldenjoy the benefits associated with employing a dynamic threshold valuefor determining whether a line includes speech.

[0058] Although the present invention has been shown and described withrespect to several preferred embodiments thereof, various changes,omissions and additions to the form and detail thereof, may be madetherein, without departing from the spirit and scope of the invention.

What is claimed is:
 1. A method for conferencing, comprising: receivingaudio signals over a plurality of ports; for at least one port,determining a dynamic threshold value based on one or morecharacteristics of signals received on the port; associating saiddynamic threshold value with the port; and comparing one or morecharacteristics of signals subsequently received on the port to thedynamic threshold value; and summing signals received over the pluralityof ports, wherein signals received on the at least one port whosecharacteristics have a specified relationship to the dynamic thresholdvalue are not contained in the sum.
 2. The method of claim 1, whereinthe dynamic threshold value is an energy level.
 3. The method of claim1, wherein the dynamic threshold value is determined based on one ormore characteristics that comprise energy level.
 4. The method of claim1, wherein the one or more characteristics of signals subsequentlyreceived on the port compared to the dynamic threshold value compriseenergy level.
 5. The method of claim 1, wherein the specifiedrelationship to the dynamic threshold value is that of being less thanthe threshold value.
 6. The method of claim 1, further comprising:identifying which ports are receiving audio signals that contain speech;and on each such identified port, transmitting a summed signal, whereinsaid summed signal does not contain signals received on that port. 7.The method of claim 1, further comprising: identifying which ports arereceiving audio signals that contain DTMF tones; and on each suchidentified port, transmitting a summed signal, wherein said summedsignal does not contain signals received on that port.
 8. The method ofclaim 7, wherein said step of identifying comprises setting a DTMFdetect bit for a signal.
 9. The method of claim 1, further comprisingpreprocessing received audio signals by decompressing the signals. 10.The method of claim 9, wherein said step of comparing one or morecharacteristics of signals subsequently received on the port to thedynamic threshold value comprises determining whether a magnitude of adecompressed audio signal is greater than said threshold value.
 11. Themethod of claim 9, wherein said step of decompressing uses μ-lawdecompression.
 12. The method of claim 9, wherein said step ofdecompressing uses A-law decompression.
 13. The method of claim 7,further comprising the step of including signals from previouslyidentified ports in said sum after those ports are no longer identifiedas receiving signals containing one or more DTMF tones.
 14. Software forconferencing, comprising: software for receiving audio signals over aplurality of ports; for at least one port, software for determining adynamic threshold value based on one or more characteristics of signalsreceived on the port; software for associating said dynamic thresholdvalue with the port; and software for comparing one or morecharacteristics of signals subsequently received on the port to thedynamic threshold value; and software for summing signals received overthe plurality of ports, wherein signals received on the at least oneport whose characteristics have a specified relationship to the dynamicthreshold value are not contained in the sum.
 15. The software of claim14, wherein the dynamic threshold value is an energy level.
 16. Thesoftware of claim 14, wherein the dynamic threshold value is determinedbased on one or more characteristics that comprise energy level.
 17. Thesoftware of claim 14, wherein the one or more characteristics of signalssubsequently received on the port compared to the dynamic thresholdvalue comprise energy level.
 18. The software of claim 14, wherein thespecified relationship to the dynamic threshold value is that of beingless than the threshold value.
 19. The software of claim 14, furthercomprising: software for identifying which ports are receiving audiosignals that contain speech; and software for, on each such identifiedport, transmitting a summed signal, wherein said summed signal does notcontain signals received on that port.
 20. The software of claim 14,further comprising: software for identifying which ports are receivingaudio signals that contain DTMF tones; and software for, on each suchidentified port, transmitting a summed signal, wherein said summedsignal does not contain signals received on that port.
 21. The softwareof claim 20, wherein said software for identifying comprises softwarefor setting a DTMF detect bit for a signal.
 22. The software of claim14, further comprising software for preprocessing received audio signalsby decompressing the signals.
 23. The software of claim 22, wherein saidsoftware for comparing one or more characteristics of signalssubsequently received on the port to the dynamic threshold valuecomprises software for determining whether a magnitude of a decompressedaudio signal is greater than said threshold value.
 24. The software ofclaim 22, wherein said software for decompressing uses μ-lawdecompression.
 25. The software of claim 22, wherein said software fordecompressing uses A-law decompression.
 26. The software of claim 20,further comprising software for including signals from previouslyidentified ports in said sum after those ports are no longer identifiedas receiving signals containing one or more DTMF tones.
 27. A system forconferencing, comprising: means for receiving audio signals over aplurality of ports; for at least one port, means for determining adynamic threshold value based on one or more characteristics of signalsreceived on the port; means for associating said dynamic threshold valuewith the port; and means for comparing one or more characteristics ofsignals subsequently received on the port to the dynamic thresholdvalue; and means for summing signals received over the plurality ofports, wherein signals received on the at least one port whosecharacteristics have a specified relationship to the dynamic thresholdvalue are not contained in the sum.
 28. The system of claim 27, whereinthe dynamic threshold value is an energy level.
 29. The system of claim27, wherein the dynamic threshold value is determined based on one ormore characteristics that comprise energy level.
 30. The system of claim27, wherein the one or more characteristics of signals subsequentlyreceived on the port compared to the dynamic threshold value compriseenergy level.
 31. The system of claim 27, wherein the specifiedrelationship to the dynamic threshold value is that of being less thanthe threshold value.
 32. The system of claim 27, further comprising:means for identifying which ports are receiving audio signals thatcontain speech; and means for, on each such identified port,transmitting a summed signal, wherein said summed signal does notcontain signals received on that port.
 33. The system of claim 27,further comprising: means for identifying which ports are receivingaudio signals that contain DTMF tones; and ‘means for, on each suchidentified port, transmitting a summed signal, wherein said summedsignal does not contain signals received on that port.
 34. The system ofclaim 33, wherein said means for identifying comprises means for settinga DTMF detect bit for a signal.
 35. The system of claim 27, furthercomprising means for preprocessing received audio signals bydecompressing the signals.
 36. The system of claim 35, wherein saidmeans for comparing one or more characteristics of signals subsequentlyreceived on the port to the dynamic threshold value comprises means fordetermining whether a magnitude of a decompressed audio signal isgreater than said threshold value.
 37. The system of claim 35, whereinsaid means for decompressing uses μ-law decompression.
 38. The system ofclaim 35, wherein said means for decompressing uses A-law decompression.39. The system of claim 33, further comprising means for includingsignals from previously identified ports in said sum after those portsare no longer identified as receiving signals containing one or moreDTMF tones.